1. Field of the Invention
The present invention relates to a data output method for temporarily accumulating received data and performing output based on the accumulated data, a data output apparatus adopting such a data output method, and a communication system including such a data output apparatus. In particular, the present invention relates to a data output method, a data output apparatus, and a communication system suitable for real-time communication such as VoIP and streaming distribution.
2. Description of Related Art
In recent years, there is an increasing use of applications that enable real-time communication such as VoIP (Voice over Internet Protocol), VoPN (Voice over Packet Network) and streaming distribution in which voice and/or video data is transmitted and received between apparatuses through a communication network such as an IP network. In particular, IP telephony that is one example of such applications is spread rapidly. In the IP telephony system, by transmitting and receiving voice data in the form of packets through an IP network, communication is realized between an apparatus at the transmitting end of packets and an apparatus at the receiving end of packets. However, in the communication through the IP network, since the arrival delay time required for transferring packets from the apparatus at the transmitting end to the apparatus at the receiving end varies depending on each packet, there is a problem of jitter. The jitter in the arrival delay time is a serious problem for IP telephony that is required to perform real-time and continuous communication. In order to solve such a problem, in the IP telephony system, the apparatus at the receiving end of packets is provided with a buffer called a “jitter buffer”. The apparatus (IP phone set) at the receiving end having the jitter buffer realizes stable voice outputted by temporarily accumulating received packets in the jitter buffer and absorbing the jitter in the arrival delay time before outputting the packets as voice.
In order to solve such a problem, in the IP telephony system, the apparatus at the receiving end of packets is provided with a buffer called a “jitter buffer”. The apparatus (IP phone set) at the receiving end having the jitter buffer realizes stable voice outputted by temporarily accumulating received packets in the jitter buffer and absorbing the jitter in the arrival delay time before outputting the packets as voice. For the jitter buffer, an initial accumulation amount to be used as a basis for starting to output the accumulated packets and an upper accumulation limit to be used as a basis for discarding the accumulated packets are set. Therefore, the apparatus at the receiving end of packets starts to accumulate the packets after starting reception of the packets, starts to output (reproduce) voice based on the packets accumulated so far at the time point the accumulated amount of packets reaches the initial accumulation amount, and discards the received packet when the accumulated amount of packets exceeds the upper accumulation limit.
When outputting voice using the jitter buffer in such a manner, the larger the accumulation capacity of the jitter buffer for accumulating packets, the greater the ability of absorbing the jitter in the arrival delay time, but there is a problem that the delay from the arrival of a packet to the output of the packet becomes larger. In order to solve such a problem, Japanese Patent Application Laid-Open No. 2003-87317 discloses a method in which an arrival delay time is calculated by measuring the arrival time intervals of packets, and a discard threshold value, namely the accumulation capacity of the jitter buffer is determined based on the jitter in the calculated arrival delay time.
Moreover, a PoC (Push-to-Talk over Cellular) service that enables the IP telephony technique to be used in a manner similar to transceivers and is capable of transmitting packets including voice data to a plurality of apparatuses in a broadcast manner has attracted the attention. The characteristics of the PoC service include a half-duplex communication in which one of a plurality of apparatuses acquires a transmission right to transmit voice data and the apparatus having the transmission right transmits voice data to the other apparatuses in a broadcast manner, and the ability to perform simultaneous communication among three or more apparatuses.
However, when the technique disclosed in Japanese Patent Application Laid-Open No. 2003-87317 that is the full duplex communication is applied to the above-mentioned PoC service of half-duplex communication in which an apparatus having the transmission right changes frequently, the following problems arise.
Specifically, immediately after the transfer of the transmission right from one apparatus to another, there is a high possibility that burst reception of packets including voice data may occur in the apparatus at the receiving end, and thus there is also a high possibility that the accumulated amount in the jitter buffer may exceed the upper accumulation limit of the apparatus at the receiving end. Consequently, since the apparatus at the receiving end has no choice but to discard a large amount of the received voice data, there arises the problem of deterioration of sound quality. Thus, for the PoC service in which the transmission right is transferred frequently among a plurality of apparatuses, the above-mentioned deterioration of sound quality is a problem that cannot be ignored.
Moreover, when the transmission right is transferred from a certain apparatus and the certain apparatus becomes the apparatus at the receiving end, the jitter buffer of the apparatus at the receiving end is empty. Therefore, until the accumulated amount of voice data received immediately after the transfer of the transmission right reaches the initial accumulation amount, the reproduction of voice based on voice data is not started, and there is a problem that a delay in voice reproduction occurs. In the PoC service in which the transmission right is transferred frequently, reproduction delay that occurs in such a situation is a problem that cannot be ignored.
Further, since a communication route until the voice data transmitted from a transmitting source apparatus reaches an apparatus at the receiving end varies depending on each transmitting source apparatus, the state of delay in the apparatus at the receiving end also varies depending on each transmitting source apparatus. Hence, for example, in a method of optimizing parameters necessary for the management of the buffer after starting to receive data as disclosed in the Japanese Patent Application Laid-Open No. 2003-87317, it is necessary to perform the process of optimizing the parameters whenever the transmission right is transferred, and consequently there arises the problem of deterioration of sound quality until the optimization of the parameters has been completed. For the PoC service in which communication among three or more apparatuses is realized and the transmission right is transferred frequently, such deterioration of sound quality is a problem that cannot be ignored.